dmance

Regarding DSD over DOP via SRC-DX. SRC-DX was only specified to support DOP DSD128 ...as this is a single SPDIF stream and the DOP mechanism is defined. The CT7601 chip used does have the internal DOP support up to DSD256 but the AES dual-date-date mode to support DBNC does not define DoP. If it does work ...its ad-hoc and fortunate. I need to ask a SRC-DX/Chord DAVE owner to try this out. *thanks*

@seeteeyou @Zaphod BeeblebroxWAV is uncompressed. If you are referring to WAVPACK, PGGB-IT! uses the FFMPEG internal code base, not the wavpack libs. So files are exact lossless but the bitstream is slightly different. FYI, this is also the case with FLAC. If your sequence is PGGB-IT! -to- WAV -to- FLAC (via XIPH exe) and compare to PGGB-IT! -to- FLAC (direct), the files will be different but the contents are lossless.

I have extensive experience in testing how sensitive a DAC is to proximate components. In a word: very. I used a RF spectrum analyzer connected to the output of a DAC (close coupled) and testing how much RF noise was passed in from input signal connections, power and radiated thru open air. My test DAC (Chord Hugo2) was on battery. Even at distances of 10m (~33') the noise from digital source components impinged on a DAC. This was due to both free air plus RF noise backing up to the AC mains in the wall from the power supply ...and hence finding its way nearer to the DAC. Putting your digital kit in a Faraday/shielded box is extreme but makes a measurable difference. Direct signals to a DAC test indicate that you can never allow any RF noise. I avoid galvanic (metal) signal at all costs. TOSLINK (up to 192k) or OPTO-DX Optical (up to 768k). Power? Use batteries if at all possible ...with added filtering to smooth out inverter noise or even lithium electrochemical discharge noise. This ends up making your audio components look somewhat like a nuclear physics science experiment... but its the only way, IMO, you can reach an endgame digital chain. Full disclosure: I am www.audiowise.ca.

When doing an apples/apples comparison its important to ensure the galvanic signal path and inputs for both is the same. You could try repeating this test by sending the PGGB file thru MScaler (it will go into bypass if the input sample rate is 16fs) and DBNC into Dave. Then do the normal MScaler upsample to 16fs and DBNC into Dave. In both cases make sure any USB cable is removed from Dave (as well as any other coax inputs you may have).

I fell in love with PGGB after upsampling the first few tracks with the demo version. After purchasing the licensed version I knew I wanted to use the command line version for better integration into my AudioWise product testing environment. I then evolved a very useful workflow utility ...something that would make selecting folders and managing batch PGGB processing easier. I decided that it was too good not to share with other PGGB users. So today I have posted, as a $free product, PGGB-IT! Workflow App on my AudioWise site. It's a Windows 10 console application that does PGGB batch processing on a queue of folders stored in a Windows Explorer library. The included companion QUEUE-IT! application allows you to add folders to the queue using per-folder drag-drop from Explorer or per-album 'Send to' directly from JRiver Media Center. PGGB-IT! then grabs the list of input folders from the library and runs PGGB (programmatically) on them to the output folder - using FQN pathing. And while you are waiting for the current batch to finish, you can load up more folders in the queue and PGGB-IT! will process those as well. The diagram below says it all If there are any Window10 users out there playing with PGGB, please download and give it a try! Dan

Chord DACs are 24 bits internal. So even a 32bit USB setting will waste 8 bits. Of course DX is just dual SPDIF which is 24 bit in the signal. As mentioned by others, set PGGB to output 24 bits for best sq to Chord DACs

I ran out of space in Dropbox for my posted PGGB'd music videos link. Use this Google Drive link here. Thanks, Dan

I stumbled upon PGGB during my weekly audio forum visits and downloaded it. After a few days, here are my comments: First ... a huge congratulations to ZB and the group that made this happen. Thanks to Ray G for (again) being involved at the forefront of digital audio and his effort to help communicate this amazing development to a wider audience. And thanks for all the fish. Introduction: I jumped on the Chord DAC bandwagon with hearing Rob Watts' large tap count sinc filters and his excellent analog section design. I sensed that Rob's approach was the right way to build a DAC and his millon tap MScaler confirmed (at least for me) that his mathematically perfect conversion could also be emotionally satisfying. Of course, digital processing is not the entire story - a DACs final D/A stage and output clocking is also subject to many and myriad influences of EMI and such. I started Audiowise Inc (www.audiowise.ca) with OPTO-DX optical isolation of the Chord DX interface and I’ve evolved a range of solutions to address the detrimental effects of RF noise. In 2019 I prodded Jussi L. to push the envelope with his HQPlayer PCM filters and noise shapers to meet or exceed MScaler sound quality. His 2M tap Sinc-L filter and 15th order LNS15 noise shaper then became my reference for up-sampling. It was clear that more detail retrieval was possible with more taps and worthwhile to adopt for playback - despite the one second delay issues. So, since early 2020 my current chain is attached, IMO, an extremely well RF isolated DAVE configured with my full kit of products that incorporate everything I’ve learned about the insidious nature of RF noise. If I had not heard PGGB, I would have been quite satisfied with the sound from HQP's 2 million taps. Now, after only a few days with PGGB, it's apparent that 2 million is not enough. Nowhere close to it. Go big or go home, I say. PGGB Usage: For PGGB installation I used a 2019 vintage Lenovo ThinkPad P71 laptop with 16Gb ram and a fast SSD. Processed WAV files were copied to my NUC and I used a remote desktop session to HQPlayer or JRiver MC for my tests. Previously, with HQP upsampling, my NUC was at 40-50% CPU usage during playback. Now, with the complete separation of the heavy lifting of upsampling away from actual playback, my NUC trickles by on 5-10% CPU ...and likely far less RF emissions. I'm quite satisfied with the PGGB user interface - mostly because ZB's 'Auto' modes are not compromised but actually provide the optimal setting. So, I set Auto taps, Auto Noise Shaping and Auto Gain. I’m also quite satisfied with the performance; however, I would like more descriptive status ...especially during periods of inactivity and, on launch, the long delay between the disappearance of the startup screen and appearance of GUI interface regularly had me concerned that PGGB had crashed. I set PGGB upsampling to 705.6/768 16fs and bit depth to 24-bits ... because that's what I used with HQPlayer and it's what MScaler's DBNC interface is. I use SRC-DX USB-to-DBNC bridge to avoid USB noise so I have not tried a 32bit file direct via USB. However, I believe USB’s 32bits is truncated to 24 at the DAVE’s signal interface anyhow. With HQP upsampling, the need to set a global -3dB (or -4dB) attenuation to avoid inter-sample overs bothered me for the reduction in volume and dynamic range. So, I am very happy that PGGB Auto Gain minimally attenuates yet perfectly avoids inter-sample overs. As the status messages whiz by I am seeing far less than -1dB of applied gain and typically less than -0.5dB so everything is louder and closer to master 0dB volume. The compromise of minor variation in gain on a per-track basis is not noticeable and, anyhow, each album is produced with a different average volume. This is no issue for me as I have the remote volume control in my hand full time and am always tweaking volume anyhow. I set TRANSPARENCY to Natural and PRESENTATION to Transparent. Regarding noise shaping...the PGGB Auto noise shaping works very well to minimize noise - I trust ZB that this is optimal ...sure sounds like it. I set the HF NOISE FILTER to 'Minimal' since, IMO, it sounds best. DAVE does its final noise shaping after its own application of an HF Filter and its excellent volume control. I processed dozens of tracks in trial mode (the first 5 tracks from each album I selected)- My playback was done manually with HQPlayer 4.10.3 and JRiver MC. I used remote desktop from my laptop to my NUC for control. I made 100% sure the audio files were untouched during playback. SQ Impressions: Stunningly good. But you might have to experience it with 16fs or better to have the full transparency reveal itself. Some tracks present with just more nuance and realism; some just sound more natural. All sound better and increase my experience of Qualia …which should be everyone’s goal. From an Audiowise perspective, PGGB is a much better tool than HQP to help me plumb further depths of sound quality and make better products. I am so glad this has been realized. Again, kudos to everyone involved. I hope Jussi L. has downloaded PGGB and hears how off-line processing with ‘practically' infinite taps sounds better than his best real-time approach. I'd like to ask ZB to report on any round-trip analysis: like redbook upconversion to a float output (WAV 32 Float?) followed by downconversion to the original. Ideally this process is transparent and helps confirm that PGGB is not adding euphonics but actually demonstrating that the reconstruction is optimal to the theoretical. If confirmed, ZB should publish a paper and secure implementation patents for having realized Nyquist-Shannon. Rob Watt’s original estimate of only 1M taps being needed for redbook roundtrip was never was publicly validated with his Davina project and now seems a tad on the low side. Workflow: A PGGB file is not stream-able today but the SQ is so compelling that I want to move ahead with a workable environment to enjoy both my own files and streamed content. WAV files are big but they have format ubiquity on playback software. I was happy to get back to using JRiver MC for playback since I have mixed feelings about Roon. Monkeys Audio is a great lossless compression program that maintains all WAV metadata. (here) and MC is a champion with full support. I converted all my PGGB'd .WAV files to .APE for seamless playback via MC. I was seeing 25-30% file sizes. That is, my typical 1000Mbyte .WAV files became ~270Mbyte .APE files. Some of this is due to my 24-bit sample leaving an empty 8-bits in WAVs 32-bit PCM word. For me, the good news about .APE compression is that the 2TB SSD in my NUC can now hold a ~7500 track PGGB music collection !! …which is plenty. FYI - the decoding from APE to PCM in MC takes about 0.5sec with prebuffering of the track. So, for the lossless purists, all PCM samples are just read from RAM during playback. ZB…PGGB has to be integrated into playback software (please!) for automatic conversion and file management. In the interim, I will work on my own workflow components for my continued testing. Video Files: Audiophiles, in general, seem to shun consuming video media with their music and both HQPlayer and ROON have stated they have no intention of adding video support. In my case, I find that watching short videos of live or studio recordings is a very compelling way to consume music. I have tested custom integrations of MC and HQP to achieve this and experimented with HQP-Pro to integrate WAVs into video files. The much-improved SQ of PGGB will now have me revisit this. I’m sure not every users audio setup is geared for video playback -> to a nice monitor and audio playback -> to a DAC ...but mine is. I’ve remixed PGGB’d WAV audio from a few 4K YouTube clips into MKV multiplexed video files. The results dramatically increase the feeling of being present : 4K video with 16fs remastered audio in perfect sync is amazing. For those interested, I have three clips being downloaded here (usage under fair use). You will need JRiver MC for playback of these files. Note that local audio playback will down-convert the audio to typically 48k so make sure MC is setup properly to preserve the full resolution PCM bitstream to your DAC. i'll keep reporting back ...definitely will purchase! Great job ZB and team! Dan

AO cannot degrade or enhance the bitstream from source to DAC. So when you say 'sound signatures' you mean how AO is modifying the radio frequency (RF) noise from the PC's CPU, memory access and various sub-systems. This RF noise leaves the PC and impacts your DAC and its tethered antennae (cables) to get inside the DAC to affect the clocking and small signal processing. So before you muck around with AO settings, do this: #1 - use a DAC that is immune to signal jitter (this would be any built in the past 10 years) #2 - use an optical signal transmission (Toslink or OPTO-DX if you have a Chord DAC) #3 - dont have the the DAC or source share the same power at all. Ideally, get off the AC mains. Put your DAC on a battery. #4 - avoid switch mode power supplies. #5 - physically distance your source components (anything digital) from your DAC. #6 - pay attention to magnetic fields generated by AC components and your household wiring grid. #7 - keep upgrading your transducers (speakers, hphones) to keep revealing more transparency as you improve above. Dan

From a purely number crunching perspective of Sinc filters and noise shapers, both MScaler and HQPlayer are very close ...with HQPlayer's SINC-L PCM filter at 2M taps (vs MScaler 1M taps) and LNS-15 15th order noise shaper (vs MScaler 11th order) being able to retrieve more details at 16fs upsampling, IMO. However that is not the entire story. If you are using USB to get the bitstream to your DAC, any RF noise accompanying the digital signal gets into the DACs clocking and small-signal processing. So a comparison of PC+HQP->DAC vs PC->MScaler->DAC also includes the variation in RF emissions from the PC doing the heavy lifting vs the MScaler. If you want a revelation in sound quality from your Chord DX DAC, your chain should be: PC+HQP->SCR-DX->DAC or, if you have the funds, PC+HQP->SRC-DX->OPTO-DX->DAC with OPTO-DX on battery power. Yes, its a chain of black boxes and wires that needs to be tucked away ... but it gets you a pragmatic solution to the elimination of RF noise (at least in the signal!).

DAC: Chord TT2 that's been RF isolated using audiowise products. Server: does not matter if it's fully RF isolated from the DAC. But use an optical signal chain.

It's all about RFI/EMI noise affecting the DAC ...more specifically its final D/A conversion from bits to volts that needs to be free of RF noise. Deviations in the DACs output clock and reference voltage will perturb the output waveform. These are caused by even impossibly small levels of RFI/EMI coming from digital devices, switch mode supplies, the AC mains, etc to impinge on the DAC. More RF noise = worse SQ and more fatigue. Our ear/brain only relents and relaxes to digital music when the DAC output is as RF noise free as possible ...well below what we can measure. Otherwise we all 'feel' the fatigue. I know all this to be true because I continue to research these boundaries and provide products and honest information. Here's a few good tips: - Use batteries to isolate the digital components from analog components. They are low noise and good sink for conducted RF noise. I recommend the Goal Zero lithium batteries - affordable and available on Amazon. - Use components with DC power inputs because this makes RF noise easier to tame. Add supplemental RF noise filters on DC power. - Use optical signal connections. (Toslink or OPTO-DX for Chord DACs). Modern DACs are 100% immune to jitter as the input and output clocks are independent. - Use highly shielded cables. For coax, look for 95dB or higher 'screening factor' from Belden. For power, look for triple shielded power from Gotham. And also go as short as possible. - Stop the pathways for RF noise at source and DAC. Use physical distance between Source and DAC components. Use aggressive RF-STOP shielding (like what I provide). Finally...Don't compromise. RF is insidious and gets in through any and all metal conductors. Your final system may look like a science project but it will provide phenomenally transparent SQ from any recording regardless of bitrate. Dan

@Racerxnet Just to frame your consideration of a solution: Please understand that nothing you do to your PC alters the fidelity of the bitstream. Any BIOS changes or other OS settings or hardware tweaks still let the digital audio get to your DAC with 100% bit perfection. If it was not the case you would hear dropouts, pops and clicks. Whenever you hear a change in the sound quality from your DAC by changing your PC is because you alter the RF emissions profile: through the signal cable, through the AC connection and radiated in free air. Your DAC is living in a sea of RF noise and each alteration of the frequency spectrum alters the DACs small signal processing in its final D/A stage. So changing the BIOS alters the amount and distribution of RF energy generated. Trying to fix your system without this basic understand leads to confusion and frustration. Try this: Get your digital source components off the same AC grid as your DAC. I would suggest putting one other on a battery for 100% isolation (not from AC mains noise but from RF noise from your PC getting through to your DAC). If your DAC is halfway competent its inputs are double-clocked and immune to jitter. I know Chord and PSAudio and many other name brands are. So if this is the case you want to use TOSLINK optical. Physically separate your PC and your DAC by as much as is practical. Get a long TOSLINK cable and extend it to the max. Your PC is radiated noise that travels across open air to your DAC (and the metal tentacles (read: cables) act as antenna to transmit and receive RF noise). Dan

Only possible to stop MHz-GHz is with a RF isolation chamber. And i don't quite understand the mechanism ...but DACs become much more transparent when they measure zero ambient RF noise.

Even the cheapest DAC decodes the bitstream without error. ...just like this CA post is coming from a distant server thru over half a continent of wires and switches. I don't see you touting using special laptops or phones to accurately decode the bits to English. That's stupid, right? I don't want to flog a dead horse but 18 months ago I walked in your shoes and believed the same things. My education came from the frustration of a CD player over spdif sounding much better than the same ripped CD on a laptop thru USB. Same DAC in both cases and its the same case you are making: system A sounds better than system B even tho the bits are the same. I made ludicrous attempts at fixing this ...magic cables, audiophile tuning the laptop, changing OS, loading the song to Ram, etc. Yes these improved/changed the laptop sound but I could never match the $100 CD player. So if you are going to believe in the invisible and intangible ...at least make it scientifically sound. Understand that the electronics in the laptop are buzzing with radio frequency noise. MHz to near GHz and these find their way across metal in the USB cable and into the DAC. These don't manifest as digital clicks and pops, of course, rather the noise mucks up the DACs analog stage final conversion so I lose things that my ear/brain needs: depth, separation, etc. So, I got the laptop to sound exactly as good as the CD by adding an intona AND 15 clamp-on ferrites to the USB cable. This squashes the noise on the sheath and the four wires in the cable. I consider this a 'solution' to the problem rather than the 'try-this/try-that' hacks that move the noise or alter its character without understanding how close to perfect you can get. So please understand that a raspberry Pi delivers the bits as reliably as a $10,000 server ...to your USB DAC ...but sounds worse because it throws huge noise into the USB. You can make the raspberry sound the same by adding an intona and $50 worth of ferrites. But nobody reading this post will believe or try this.